baresip: tls: accept error: no shared cipher

I’m using baresip v3.1.0 to connect to my Twilio account. I can place calls from baresip successfully, but when I receive a call, I see the following error message:

tls: accept error: (r=-1, ssl_err=1) tls: 139931151087424:error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher:…/ssl/statem/statem_srvr.c:2283:

After seeing this announcement from Twilio, I’ve upgraded the openssl library to the latest one (3.1.0) like this:

(this is from my Dockerfile)

    RUN git clone -b v3.1.0 --single-branch https://github.com/baresip/baresip.git && \
        cd baresip && \
        cmake -B build -DCMAKE_BUILD_TYPE=RelWithDebInfo -DCMAKE_C_COMPILER=clang-11 \
            -DCMAKE_CXX_COMPILER=clang++-11 -DCMAKE_INSTALL_PREFIX=/usr -DOPENSSL_ROOT_DIR=/install/openssl/openssl-3.1.0

But to no avail. I still get the same error, so I suspect there something else wrong.

Do you have any idea where I should be looking?

This is my baresip config file:

#
# baresip configuration -- example for linux
#

#------------------------------------------------------------------------------

# Core
poll_method		epoll		# poll, select, epoll ..

# SIP
#sip_listen		0.0.0.0:5060
#sip_certificate	cert.pem
#sip_cafile		/etc/ssl/certs/ca-certificates.crt
sip_transports		tls 
#udp,tcp,tls,ws,wss
#sip_trans_def		udp
sip_verify_server	no
#sip_tos			160 # See TOS fields!

## TOS fields ##
#    7     6     5     4     3     2     1     0
# +-----+-----+-----+-----+-----+-----+-----+-----+
# |   PRECEDENCE    |          TOS          | MBZ |
# +-----+-----+-----+-----+-----+-----+-----+-----+
# TOS:
# bit 4 - low delay
# bit 3 - high throughput
# bit 2 - high reliability
# bit 1 - low cost
# MBZ:
# bit 0 - zero

# Call
call_local_timeout	120
call_max_calls		4
call_hold_other_calls	yes

# Audio
#audio_path		/usr/share/baresip
audio_player		pulse
audio_source		pulse
audio_alert		pulse,speaker
#ausrc_srate		48000
#auplay_srate		48000
#ausrc_channels		0
#auplay_channels	0
#audio_txmode		poll		# poll, thread
audio_level		no
ausrc_format		s16		# s16, float, ..
auplay_format		s16		# s16, float, ..
auenc_format		s16		# s16, float, ..
audec_format		s16		# s16, float, ..
audio_buffer		20-160		# ms
audio_telev_pt		101		# payload type for telephone-event

# Video
#video_source		v4l2,/dev/video0
#video_display		x11,nil
video_size		640x480
video_bitrate		1000000
video_fps		30.00
video_fullscreen	yes
videnc_format		yuv420p

# AVT - Audio/Video Transport
rtp_tos			184
rtp_video_tos		136
#rtp_ports		10000-20000
#rtp_bandwidth		512-1024 # [kbit/s]
rtcp_mux		no
jitter_buffer_type	fixed		# off, fixed, adaptive
jitter_buffer_delay	5-10		# frames
#jitter_buffer_wish	6		# frames for start
rtp_stats		no
#rtp_timeout		60

# Network
#dns_server		1.1.1.1:53
#dns_server		1.0.0.1:53
#dns_fallback		8.8.8.8:53
#net_interface		eth0
# Play tones
#file_ausrc		aufile
#file_srate		16000
#file_channels		1

#------------------------------------------------------------------------------
# Modules

module_path		/usr/lib/x86_64-linux-gnu/baresip/modules

# UI Modules
module			stdio.so
#module			cons.so
#module			evdev.so
#module			httpd.so

# Audio codec Modules (in order)
module			opus.so
#module			amr.so
#module			g7221.so
#module			g722.so
#module			g726.so
#module			g711.so
#module			gsm.so
#module			l16.so
#module			mpa.so
#module			codec2.so
#module			ilbc.so

# Audio filter Modules (in encoding order)
module			auconv.so
module			auresamp.so
#module			vumeter.so
#module			sndfile.so
#module			plc.so
#module			webrtc_aec.so

# Audio driver Modules
#module			alsa.so
module			pulse.so
#module			jack.so
#module			portaudio.so
#module			aubridge.so
#module			aufile.so
#module			ausine.so

# Video codec Modules (in order)
#module			avcodec.so
#module			vp8.so
#module			vp9.so

# Video filter Modules (in encoding order)
#module			selfview.so
#module			snapshot.so
#module			swscale.so
#module			vidinfo.so
#module			avfilter.so

# Video source modules
#module			v4l2.so
#module			vidbridge.so

# Video display modules
#module			directfb.so
#module			x11.so
#module			sdl.so
#module			fakevideo.so

# Audio/Video source modules
#module			avformat.so
#module			gst.so
#module			gst_video.so

# Compatibility modules
#module			ebuacip.so
module			uuid.so

# Media NAT modules
module			stun.so
module			turn.so
module			ice.so
#module			natpmp.so
#module			pcp.so

# Media encryption modules
module			srtp.so
#module			dtls_srtp.so
#module			zrtp.so


#------------------------------------------------------------------------------
# Application Modules

module_app		account.so
#module_app		contact.so
#module_app		debug_cmd.so
#module_app		echo.so
#module_app		gtk.so
module_app		menu.so
#module_app		mwi.so
#module_app		presence.so
#module_app		serreg.so
#module_app		syslog.so
module_app		mqtt.so
#module_app		ctrl_tcp.so
#module_app		ctrl_dbus.so
#module_app		httpreq.so
#module_app		multicast.so
module_app		netroam.so


#------------------------------------------------------------------------------
# Module parameters

# DTLS SRTP parameters
#dtls_srtp_use_ec	prime256v1

# UI Modules parameters
cons_listen		0.0.0.0:5555 # cons - Console UI UDP/TCP sockets

http_listen		0.0.0.0:8000 # httpd - HTTP Server

ctrl_tcp_listen		0.0.0.0:4444 # ctrl_tcp - TCP interface JSON

evdev_device		/dev/input/event0

# Opus codec parameters
opus_bitrate		48000 # 6000-510000
opus_stereo		yes
#opus_sprop_stereo	yes
#opus_cbr		no
#opus_inbandfec		no
#opus_dtx		no
#opus_mirror		no
#opus_complexity	10
#opus_application	audio	# {voip,audio}
opus_samplerate	48000
#opus_packet_loss	10	# 0-100 percent (expected packet loss)

# Opus Multistream codec parameters
#opus_ms_channels	2	#total channels (2 or 4)
#opus_ms_streams	2	#number of streams
#opus_ms_c_streams	2	#number of coupled streams

vumeter_stderr		yes

#jack_connect_ports	yes

# Selfview
video_selfview		window # {window,pip}
#selfview_size		64x64

# ZRTP
#zrtp_hash		no  # Disable SDP zrtp-hash (not recommended)

# Menu
#redial_attempts	0 # Num or <inf>
#redial_delay		5 # Delay in seconds
#ringback_disabled	no
#statmode_default	off
#menu_clean_number	no
#sip_autoanswer_method	rfc5373 # {rfc5373,call-info,alert-info}
#ring_aufile		ring.wav
#callwaiting_aufile	callwaiting.wav
#ringback_aufile	ringback.wav
#notfound_aufile	notfound.wav
#busy_aufile		busy.wav
#error_aufile		error.wav
#sip_autoanswer_aufile	autoanswer.wav

# GTK
#gtk_clean_number	no

# avcodec
#avcodec_h264enc	libx264
#avcodec_h264dec	h264
#avcodec_h265enc	libx265
#avcodec_h265dec	hevc
#avcodec_hwaccel	vaapi
#avcodec_profile_level_id 42002a

# ctrl_dbus
#ctrl_dbus_use	system		# system, session

# mqtt
#mqtt_broker_host	sollentuna.example.com
#mqtt_broker_port	1883
#mqtt_broker_cafile	/path/to/broker-ca.crt	# set this to enforce TLS
#mqtt_broker_clientid	baresip01	# has to be unique
#mqtt_broker_user	user
#mqtt_broker_password	pass
#mqtt_basetopic		baresip/01

# sndfile
#snd_path		/tmp

# EBU ACIP
#ebuacip_jb_type	fixed	# auto,fixed

# HTTP request module
#httpreq_ca		trusted1.pem
#httpreq_ca		trusted2.pem
#httpreq_dns		1.1.1.1
#httpreq_dns		8.8.8.8
#httpreq_hostname	myserver
#httpreq_cert		cert.pem
#httpreq_key		key.pem

# multicast receivers (in priority order)- port number must be even
#multicast_call_prio	0
#multicast_ttl		1
#multicast_jbuf_type	fixed		# off, fixed, adaptive
#multicast_jbuf_delay	5-10		# frames
#multicast_jbuf_wish	6		# frames for start
#multicast_listener	224.0.2.21:50000
#multicast_listener	224.0.2.21:50002

# avformat
#avformat_pass_through	yes